Download A Csound Opcode for a Triode Stage of a Vacuum Tube Amplifier The Csound audio programming language adheres to the inputoutput paradigm and provides a large number of specialized commands (called opcodes) for processing output signals from input signals. Therefore it is not directly suitable for component modeling of analog circuitry. This contribution describes an attempt to virtual analog modeling and presents a Csound opcode for a triode stage of a vacuum tube amplifier. Externally it communicates with other opcodes via input and output signals at the sample rate. Internally it uses an established wave digital filter model of a standard triode. The opcode is available as library module.
Download Time-Domain Chroma Extraction In this paper, a novel chroma extraction technique called TimeDomain Chroma Extraction (TDCE) is introduced. In comparison to many other known schemes, the calculation of a time-frequency representation is unnecessary since the TDCE is a pure sample-bysample technique. It mainly consists of a pitch tracking module that is implemented with a phase-locked loop (PLL). A set of 24 bandpass filters over two octaves is designed with the F 0 output of the pitch tracker to estimate a chroma vector. To verify the performance of the TDCE, a simple chord recognition algorithm is applied to the chroma output. The experimental results show that this novel time-domain chroma extraction technique yields good results while requiring only minor complexity and thus, enables the extraction of musical features in real-time on low-cost DSP platforms.
Download Comparison of Various Predictors for Audio Extrapolation In this study, receiver-based audio error concealment in the context of low-latency Audio over IP transmission is analyzed. Therefore, the well-known technique of audio extrapolation is investigated concerning its usability in real-time scenarios, its applied prediction techniques and various transmission parameters. A large-scale automated evaluation with PEAQ and a MUSHRA listening test reveal the performance of the various extrapolation setups. The results show the suitability of extrapolation to perform audio error concealment in real-time and the qualitative superiority of block based methods over sample based methods.
Download Physical Modeling of the MXR Phase 90 Guitar Effect Pedal In this study, a famous boxed effect pedal, also called stompbox, for electrical guitars is analyzed and simulated. The nodal DK method is used to create a non-linear state-space system with Matlab as a physical model for the MXR Phase 90 guitar effect pedal. A crucial component of the effect are Junction Field Effect Transistors (JFETs) which are used as variable resistors to dynamically vary the phase-shift characteristic of an allpass-filter cascade. So far, virtual analog modeling in the context of audio has mainly been applied to diode-clippers and vacuum tube circuits. This work shows an efficient way of describing the nonlinear behavior of JFETs, which are wide-spread in audio devices. To demonstrate the applicability of the proposed physical model, a real-time VST audio plug-in was implemented.
Download Low-Delay Error Concealment with Low Computational Overhead for Audio over IP Applications A major problem in low-latency Audio over IP transmission is the unpredictable impact of the underlying network, leading to jitter and packet loss. Typically, error concealment strategies are employed at the receiver to counteract audible artifacts produced by missing audio data resulting from the mentioned network characteristics. Known concealment methods tend to achieve only unsatisfactory audio quality or cause high computational costs. Hence, this study aims at finding a new low-cost concealment strategy using simplest algorithms. The proposed system basically consists of an period extraction and alignment module to synthesize concealment signals from previous data. The audio quality is evaluated in form of automated measurements using PEAQ. Furthermore, the system’s complexity is analyzed by drawing the computational costs of all required modules in all operating modes and comparing its computational load versus another concealment method based on auto-regressive modeling.
Download Feature design for the classification of audio effect units by input/output measurements Virtual analog modeling is an important field of digital audio signal processing. It allows to recreate the tonal characteristics of real-world sound sources or to impress the specific sound of a certain analog device upon a digital signal on a software basis. Automatic virtual analog modeling using black-box system identification based on input/output (I/O) measurements is an emerging approach, which can be greatly enhanced by specific pre-processing methods suggesting the best-fitting model to be optimized in the actual identification process. In this work, several features based on specific test signals are presented allowing to categorize instrument effect units into classes of effects, like distortion, compression, modulations and similar categories. The categorization of analog effect units is especially challenging due to the wide variety of these effects. For each device, I/O measurements are performed and a set of features is calculated to allow the classification. The features are computed for several effect units to evaluate their applicability using a basic classifier based on pattern matching.
Download Low-delay vector-quantized subband ADPCM coding Several modern applications require audio encoders featuring low data rate and lowest delays. In terms of delay, Adaptive Differential Pulse Code Modulation (ADPCM) encoders are advantageous compared to block-based codecs due to their instantaneous output and therefore preferred in time-critical applications. If the the audio signal transport is done block-wise anyways, as in Audio over IP (AoIP) scenarios, additional advantages can be expected from block-wise coding. In this study, a generalized subband ADPCM concept using vector quantization with multiple realizations and configurations is shown. Additionally, a way of optimizing the codec parameters is derived. The results show that for the cost of small algorithmic delays the data rate of ADPCM can be significantly reduced while obtaining a similar or slightly increased perceptual quality. The largest algorithmic delay of about 1 ms at 44.1 kHz is still smaller than the ones of well-known low-delay codecs.
Download Cascaded prediction in ADPCM codec structures The aim of this study is to demonstrate how ADPCM-based codec structures can be improved using cascaded prediction. The advantage of predictor cascades is to allow the adaption to several signal conditions, as it is done in block-based perceptual codecs like MP3, AAC, etc. In other words, additional predictors with a small order are supposed to enhance the prediction of non-stationary signals. The predictor cascade is complemented with a simple adaptive quantizer to yield a simple exemplary codec which is used to demonstrate the influence of the predictor cascade. Several cascade configurations are considered and optimized using a genetic algorithm. A measurement of the prediction gain and the ODG score utilizing the PEAQ algorithm applied to the SQAM dataset shall reveal the potential improvements.
Download Downmix compatible conversion from mono to stereo in time- and frequency-domain Even in a time of surround and 3D sound, many tracks and recordings are still only available in mono or it is not feasible to record a source with multiple microphones for several reasons. In these cases, a pseudo stereo conversion of mono signals can be a useful preprocessing step and/or an enhancing audio effect. The conversion proposed in this paper is designed to deliver a neutral sounding stereo image by avoiding timbral coloration or reverberation. Additionally, the resulting stereo signal is downmix-compatible and allows to revert to the original mono signal by a simple summation of the left and right channels. Several configuration parameters are shown to control the stereo panorama. The algorithm can be implemented in time-domain or also in the frequency-domain with additional features, like center focusing.
Download Signal-Matched Power-Complementary Cross-Fading and Dry-Wet Mixing The blending of audio signals, called cross-fading, is a very common task in audio signal processing. Therefore, digital audio workstations offer several fading curves to select from. The choice of the fading curve typically depends on the signal characteristics and is supposed to result in a mixed signal featuring power and loudness close to the input signals. This work derives a correlationbased design of the fading curves to achieve exact power consistency to avoid audible fluctuations of the signal’s loudness. This principle is extended to the problem of mixing original signals with effect-processed signals using the dry-wet balance. Weighting coefficients for dry and wet signals are derived which realize the desired dry-wet balance but maintain the signal power.